There is currently a great interest world wide in providing inter-working between Telephony and Internet Protocol (IP) based networks in order to extend their respective services and advantages into the other network. One of the main reasons behind this interest resides on the increased flexibility and reduced operating cost characteristics of IP-based networks as transporting circuit switched network related signalling information between signalling points. Such inter-working between Telephony and IP-based networks is commonly represented by an inter-working node acting as the border between both corresponding domains, the Telephony domain and the IP domain. This inter-working node is in charge of attending all the incoming requests from the IP domain as well as of sending all the traffic coming from the telephony domain to the IP domain.
One of the preferred protocols in the IP domain for call/session control is the Session Initiation Protocol (SIP), which is now under specification by the SIP Working Group of the Internet Engineering Task Forces (SIP IETF WG), within the Transport Area. In fact, several SIP entities, the so called Call Status Control Function (CSCF), have been defined in the third Generation Partnership Project (3GPP) which allow the Circuit Switched and the IP multimedia domains be interconnected.
In this respect, it is noticeable the effort in order to define protocol mapping mechanisms to make this inter-working possible between IP and Circuit Switched networks. For example, the SIPPING Working Group within the Transport Area of the IETF defines the SIP-T framework to facilitate the interconnection of the Public Switched Telephone Network (PSTN) with the IP network. On the other hand, the Integrated Services Digital Network (ISDN) is nowadays a world wide spread network shared by both fixed and mobile networks wherein the ISDN User Part (ISUP) of a Signalling System #7 (SS7) is the signalling protocol that said ISDN makes use of. In this respect, the ISUP to SIP Mapping is another initiative from the SIP IETF WG, describing a way to perform the mapping between said two signalling protocols.
The current architectural proposals for this inter-working node, border between the Telephony and the IP domains, go towards some gateway decomposition approach, as basically resulting from studies carried out by the International Telecommunications Union (ITU), the European Telecommunication Standard Institute (ETSI), the Multi-Switching Service Forum (MSF), and the above referred IETF standardization bodies.
The current inter-working node comprises a Media Gateway (MGW) responsible for establishing call connections over a bearer network, and a Media Gateway Controller (MGC) implementing call control related procedures connected to the Media Gateway. Both nodes communicate to each other by making use of a control protocol, which is described in the ITU-T Recommendation H.248, and is also followed up by the MEGACO Working Group within the Transport Area in the IETF.
From a SIP domain viewpoint, the inter-working node acts as a SIP User Agent (hereinafter SIP-UA) that is connected to at least one of a plurality of SIP proxies located in the IP domain. Said inter-working between SIP-UA and SIP-Proxy is then regarded as the main bottleneck between both Telephony and IP Domains, namely between both Telephony and IP networks where SIP is used. Moreover, this main drawback extends beyond the inter-working between Telephony and IP networks, This main drawback is a rather negative influence on any general scenario comprising SIP-UA and SIP-Proxy connections. In such a scenario, there is still a lack of reliable flow control mechanisms to avoid the inter-working nodes becoming temporarily unavailable, or even being completely down, just because they are not able to handle as much traffic as they might be receiving at a certain time.